mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-19 06:27:30 +00:00
108 lines
3.4 KiB
C++
108 lines
3.4 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include <assert.h>
|
|
|
|
#include <algorithm> // Access to min.
|
|
|
|
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
|
|
|
|
namespace webrtc {
|
|
|
|
size_t SyncBuffer::FutureLength() const {
|
|
return Size() - next_index_;
|
|
}
|
|
|
|
void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
|
|
size_t samples_added = append_this.Size();
|
|
AudioMultiVector::PushBack(append_this);
|
|
AudioMultiVector::PopFront(samples_added);
|
|
if (samples_added <= next_index_) {
|
|
next_index_ -= samples_added;
|
|
} else {
|
|
// This means that we are pushing out future data that was never used.
|
|
// assert(false);
|
|
// TODO(hlundin): This assert must be disabled to support 60 ms frames.
|
|
// This should not happen even for 60 ms frames, but it does. Investigate
|
|
// why.
|
|
next_index_ = 0;
|
|
}
|
|
dtmf_index_ -= std::min(dtmf_index_, samples_added);
|
|
}
|
|
|
|
void SyncBuffer::PushFrontZeros(size_t length) {
|
|
InsertZerosAtIndex(length, 0);
|
|
}
|
|
|
|
void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
|
|
position = std::min(position, Size());
|
|
length = std::min(length, Size() - position);
|
|
AudioMultiVector::PopBack(length);
|
|
for (size_t channel = 0; channel < Channels(); ++channel) {
|
|
channels_[channel]->InsertZerosAt(length, position);
|
|
}
|
|
if (next_index_ >= position) {
|
|
// We are moving the |next_index_| sample.
|
|
set_next_index(next_index_ + length); // Overflow handled by subfunction.
|
|
}
|
|
if (dtmf_index_ > 0 && dtmf_index_ >= position) {
|
|
// We are moving the |dtmf_index_| sample.
|
|
set_dtmf_index(dtmf_index_ + length); // Overflow handled by subfunction.
|
|
}
|
|
}
|
|
|
|
void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
|
|
size_t length,
|
|
size_t position) {
|
|
position = std::min(position, Size()); // Cap |position| in the valid range.
|
|
length = std::min(length, Size() - position);
|
|
AudioMultiVector::OverwriteAt(insert_this, length, position);
|
|
}
|
|
|
|
void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
|
|
size_t position) {
|
|
ReplaceAtIndex(insert_this, insert_this.Size(), position);
|
|
}
|
|
|
|
size_t SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
|
|
int16_t* output) {
|
|
if (!output) {
|
|
assert(false);
|
|
return 0;
|
|
}
|
|
size_t samples_to_read = std::min(FutureLength(), requested_len);
|
|
ReadInterleavedFromIndex(next_index_, samples_to_read, output);
|
|
next_index_ += samples_to_read;
|
|
return samples_to_read;
|
|
}
|
|
|
|
void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
|
|
end_timestamp_ += increment;
|
|
}
|
|
|
|
void SyncBuffer::Flush() {
|
|
Zeros(Size());
|
|
next_index_ = Size();
|
|
end_timestamp_ = 0;
|
|
dtmf_index_ = 0;
|
|
}
|
|
|
|
void SyncBuffer::set_next_index(size_t value) {
|
|
// Cannot set |next_index_| larger than the size of the buffer.
|
|
next_index_ = std::min(value, Size());
|
|
}
|
|
|
|
void SyncBuffer::set_dtmf_index(size_t value) {
|
|
// Cannot set |dtmf_index_| larger than the size of the buffer.
|
|
dtmf_index_ = std::min(value, Size());
|
|
}
|
|
|
|
} // namespace webrtc
|